1. Field of the Invention
The present invention relates generally to a call processing message converter and message converting method in Internet protocol (IP) telephony exchange system, which is configured in a is manner that it can control IP terminals like legacy terminals by transceiving user information with IP terminals.
2. Description of the Related Art
Recently, as the Internet is expanding rapidly to the vast majority of the world, and to meet the needs for a variety of different services, IP has been developed at a remarkable speed in terms of performances and services it provides to users. Not being satisfied here, users are continually asking for more diverse services. One of the examples is transmission of voice signals using IP network (or VoIP: Voice over Internet Protocol). Besides the data transmission of voice signals through such an IP network, other manifold services associated with the voice signal transmission, yet being a major part of the IP network, have been requested more and more.
As an attempt to meet the request, a technique for integrating generally used legacy telecommunication and VoIP is under development for interworking with a current communication network. For instance, integration of IP-based private automatic branch exchanges, i.e., IP-PBX (Internet Protocol PBX (private branch exchange)) and IP-Centrex, is the typical one. Therefore, to satisfy different needs of users, the terminals for use of an IP network should be able to have the same format and same performances with those used for PSTN (Public Switching Telephone Network).
A general IP-phone protocol currently being used right now is one of H.323 recommended by ITU-T (telecommunication standardization sector of the International Telecommunication Union), MGCP (Media Gateway Control Protocol) used between a media gateway and a media gateway controller for controlling the media gateway, or SIP (Session Initiation Protocol), a multimedia communication standard supporting integration of data, voice and image.
Normally, IP network and PSTN are separated from each other, and therefore, terminals that are accessible to each network have different protocols and different characteristics. More specifically, terminals that are connectable to PSTN are subscriber terminals including analog telephones, digital telephones or modems, while terminals that are connectable to trunk connection part include E1/T1 (E1 is a European subscriber line, a type of a dedicated circuit for transmitting data at a speed of 2.048 Mbps (megabits per second) in a manner of wide area digital transmitting technique; T1 being a digital transmission standard in for example North America that carries a digital signal level-1 (DS1) ), PRI (primary rate interface), loop and No. 7 signaling and so forth. Further, terminals that are connectable to the IP network include H.323 terminals, MGCP terminals, SIP terminals and so on. To transmit/receive (transmit and receive) voices using an IP-terminal and IP network, the terminal should be connected to an IP-line through LAN (local area network) and gatekeeper.
In other words, to enable IP network subscribers and PSTN subscribers to communicate with each other by interworking of IP network and PSTN, any type of gateway is necessary, and to seize such a gateway, an independent system for seizing internetwork should be configured. This means that IP network and PSTN have their own mutually independent number systems and terminal management systems.
In case a general legacy terminal accommodated to PSTN calls another general legacy terminal, depending on the address of a called party's terminal (domestic, long-distance, overseas, wire/wireless), a pre-designated prefix is dialed to make the call. However, if the called party uses an IP terminal, it was essential to dial a prefix seizing a gateway that is connected to a corresponding exchange.
Similarly, uses having IP terminal connected over the Internet should dial a pre-designated prefix in conforming to the address of the other party (domestic or other areas). Further, if it is necessary to make a call to the PSTN, users must dial a prefix for seizing a gateway that is connected to an exchange system to control the other party's terminal.
However, according to the conventional technology described above, mutually independent systems are in charge of controlling calls between PSTN based terminals and IP network based terminals. Thus, to configure voice channels between IP terminals and legacy terminals, two separate systems are required. This means that a service should be carried out independently over the interwork of the two systems, and integrated function and terminal management system over internetwork is therefore impossible. Unfortunately, this made it very difficult to integrate IP terminals in areas using the existing PSTN or within a building. That is, implementation of an IP based internal private automatic branch exchange got very complicated since PBX cannot process calls from IP terminals like extension telephones, and as a result thereof, a number of services provided by PBX were not available.